comparison audioio/AudioGenerator.cpp @ 0:cd5d7ff8ef38

* Reorganising code base.  This revision will not compile.

+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    Sonic Visualiser
+    An audio file viewer and annotation editor.
+    Centre for Digital Music, Queen Mary, University of London.
+    This file copyright 2006 Chris Cannam.
+    
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "AudioGenerator.h"
+
+#include "base/TempDirectory.h"
+#include "base/PlayParameters.h"
+#include "base/PlayParameterRepository.h"
+#include "base/Pitch.h"
+#include "base/Exceptions.h"
+
+#include "model/NoteModel.h"
+#include "model/DenseTimeValueModel.h"
+#include "model/SparseOneDimensionalModel.h"
+
+#include "plugin/RealTimePluginFactory.h"
+#include "plugin/RealTimePluginInstance.h"
+#include "plugin/PluginIdentifier.h"
+#include "plugin/PluginXml.h"
+#include "plugin/api/alsa/seq_event.h"
+
+#include <iostream>
+#include <math.h>
+
+#include <QDir>
+#include <QFile>
+
+const size_t
+AudioGenerator::m_pluginBlockSize = 2048;
+
+QString
+AudioGenerator::m_sampleDir = "";
+
+//#define DEBUG_AUDIO_GENERATOR 1
+
+AudioGenerator::AudioGenerator() :
+    m_sourceSampleRate(0),
+    m_targetChannelCount(1)
+{
+    connect(PlayParameterRepository::getInstance(),
+            SIGNAL(playPluginIdChanged(const Model *, QString)),
+            this,
+            SLOT(playPluginIdChanged(const Model *, QString)));
+
+    connect(PlayParameterRepository::getInstance(),
+            SIGNAL(playPluginConfigurationChanged(const Model *, QString)),
+            this,
+            SLOT(playPluginConfigurationChanged(const Model *, QString)));
+}
+
+AudioGenerator::~AudioGenerator()
+{
+}
+
+bool
+AudioGenerator::canPlay(const Model *model)
+{
+    if (dynamic_cast<const DenseTimeValueModel *>(model) ||
+	dynamic_cast<const SparseOneDimensionalModel *>(model) ||
+	dynamic_cast<const NoteModel *>(model)) {
+	return true;
+    } else {
+	return false;
+    }
+}
+
+bool
+AudioGenerator::addModel(Model *model)
+{
+    if (m_sourceSampleRate == 0) {
+
+	m_sourceSampleRate = model->getSampleRate();
+
+    } else {
+
+	DenseTimeValueModel *dtvm =
+	    dynamic_cast<DenseTimeValueModel *>(model);
+
+	if (dtvm) {
+	    m_sourceSampleRate = model->getSampleRate();
+	    return true;
+	}
+    }
+
+    RealTimePluginInstance *plugin = loadPluginFor(model);
+    if (plugin) {
+        QMutexLocker locker(&m_mutex);
+        m_synthMap[model] = plugin;
+        return true;
+    }
+
+    return false;
+}
+
+void
+AudioGenerator::playPluginIdChanged(const Model *model, QString)
+{
+    if (m_synthMap.find(model) == m_synthMap.end()) return;
+    
+    RealTimePluginInstance *plugin = loadPluginFor(model);
+    if (plugin) {
+        QMutexLocker locker(&m_mutex);
+        delete m_synthMap[model];
+        m_synthMap[model] = plugin;
+    }
+}
+
+void
+AudioGenerator::playPluginConfigurationChanged(const Model *model,
+                                               QString configurationXml)
+{
+//    std::cerr << "AudioGenerator::playPluginConfigurationChanged" << std::endl;
+
+    if (m_synthMap.find(model) == m_synthMap.end()) {
+        std::cerr << "AudioGenerator::playPluginConfigurationChanged: We don't know about this plugin" << std::endl;
+        return;
+    }
+
+    RealTimePluginInstance *plugin = m_synthMap[model];
+    if (plugin) {
+        PluginXml(plugin).setParametersFromXml(configurationXml);
+    }
+}
+
+QString
+AudioGenerator::getDefaultPlayPluginId(const Model *model)
+{
+    const SparseOneDimensionalModel *sodm =
+        dynamic_cast<const SparseOneDimensionalModel *>(model);
+    if (sodm) {
+        return QString("dssi:%1:sample_player").
+            arg(PluginIdentifier::BUILTIN_PLUGIN_SONAME);
+    }
+
+    const NoteModel *nm = dynamic_cast<const NoteModel *>(model);
+    if (nm) {
+        return QString("dssi:%1:sample_player").
+            arg(PluginIdentifier::BUILTIN_PLUGIN_SONAME);
+    }  
+    
+    return "";
+}
+
+QString
+AudioGenerator::getDefaultPlayPluginConfiguration(const Model *model)
+{
+    QString program = "";
+
+    const SparseOneDimensionalModel *sodm =
+        dynamic_cast<const SparseOneDimensionalModel *>(model);
+    if (sodm) {
+        program = "tap";
+    }
+
+    const NoteModel *nm = dynamic_cast<const NoteModel *>(model);
+    if (nm) {
+        program = "piano";
+    }
+
+    if (program == "") return "";
+
+    return
+        QString("<plugin configuration=\"%1\" program=\"%2\"/>")
+        .arg(XmlExportable::encodeEntities
+             (QString("sampledir=%1")
+              .arg(PluginXml::encodeConfigurationChars(getSampleDir()))))
+        .arg(XmlExportable::encodeEntities(program));
+}    
+
+QString
+AudioGenerator::getSampleDir()
+{
+    if (m_sampleDir != "") return m_sampleDir;
+
+    try {
+        m_sampleDir = TempDirectory::getInstance()->getSubDirectoryPath("samples");
+    } catch (DirectoryCreationFailed f) {
+        std::cerr << "WARNING: AudioGenerator::getSampleDir: Failed to create "
+                  << "temporary sample directory" << std::endl;
+        m_sampleDir = "";
+        return "";
+    }
+
+    QDir sampleResourceDir(":/samples", "*.wav");
+
+    for (unsigned int i = 0; i < sampleResourceDir.count(); ++i) {
+
+        QString fileName(sampleResourceDir[i]);
+        QFile file(sampleResourceDir.filePath(fileName));
+
+        if (!file.copy(QDir(m_sampleDir).filePath(fileName))) {
+            std::cerr << "WARNING: AudioGenerator::getSampleDir: "
+                      << "Unable to copy " << fileName.toStdString()
+                      << " into temporary directory \""
+                      << m_sampleDir.toStdString() << "\"" << std::endl;
+        }
+    }
+
+    return m_sampleDir;
+}
+
+void
+AudioGenerator::setSampleDir(RealTimePluginInstance *plugin)
+{
+    plugin->configure("sampledir", getSampleDir().toStdString());
+} 
+
+RealTimePluginInstance *
+AudioGenerator::loadPluginFor(const Model *model)
+{
+    QString pluginId, configurationXml;
+
+    PlayParameters *parameters =
+	PlayParameterRepository::getInstance()->getPlayParameters(model);
+    if (parameters) {
+        pluginId = parameters->getPlayPluginId();
+        configurationXml = parameters->getPlayPluginConfiguration();
+    }
+
+    if (pluginId == "") {
+        pluginId = getDefaultPlayPluginId(model);
+        configurationXml = getDefaultPlayPluginConfiguration(model);
+    }
+
+    if (pluginId == "") return 0;
+
+    RealTimePluginInstance *plugin = loadPlugin(pluginId, "");
+    if (!plugin) return 0;
+
+    if (configurationXml != "") {
+        PluginXml(plugin).setParametersFromXml(configurationXml);
+    }
+
+    if (parameters) {
+        parameters->setPlayPluginId(pluginId);
+        parameters->setPlayPluginConfiguration(configurationXml);
+    }
+
+    return plugin;
+}
+
+RealTimePluginInstance *
+AudioGenerator::loadPlugin(QString pluginId, QString program)
+{
+    RealTimePluginFactory *factory =
+	RealTimePluginFactory::instanceFor(pluginId);
+    
+    if (!factory) {
+	std::cerr << "Failed to get plugin factory" << std::endl;
+	return false;
+    }
+	
+    RealTimePluginInstance *instance =
+	factory->instantiatePlugin
+	(pluginId, 0, 0, m_sourceSampleRate, m_pluginBlockSize, m_targetChannelCount);
+
+    if (!instance) {
+	std::cerr << "Failed to instantiate plugin " << pluginId.toStdString() << std::endl;
+        return 0;
+    }
+
+    setSampleDir(instance);
+
+    for (unsigned int i = 0; i < instance->getParameterCount(); ++i) {
+        instance->setParameterValue(i, instance->getParameterDefault(i));
+    }
+    std::string defaultProgram = instance->getProgram(0, 0);
+    if (defaultProgram != "") {
+//        std::cerr << "first selecting default program " << defaultProgram << std::endl;
+        instance->selectProgram(defaultProgram);
+    }
+    if (program != "") {
+//        std::cerr << "now selecting desired program " << program.toStdString() << std::endl;
+        instance->selectProgram(program.toStdString());
+    }
+    instance->setIdealChannelCount(m_targetChannelCount); // reset!
+
+    return instance;
+}
+
+void
+AudioGenerator::removeModel(Model *model)
+{
+    SparseOneDimensionalModel *sodm =
+	dynamic_cast<SparseOneDimensionalModel *>(model);
+    if (!sodm) return; // nothing to do
+
+    QMutexLocker locker(&m_mutex);
+
+    if (m_synthMap.find(sodm) == m_synthMap.end()) return;
+
+    RealTimePluginInstance *instance = m_synthMap[sodm];
+    m_synthMap.erase(sodm);
+    delete instance;
+}
+
+void
+AudioGenerator::clearModels()
+{
+    QMutexLocker locker(&m_mutex);
+    while (!m_synthMap.empty()) {
+	RealTimePluginInstance *instance = m_synthMap.begin()->second;
+	m_synthMap.erase(m_synthMap.begin());
+	delete instance;
+    }
+}    
+
+void
+AudioGenerator::reset()
+{
+    QMutexLocker locker(&m_mutex);
+    for (PluginMap::iterator i = m_synthMap.begin(); i != m_synthMap.end(); ++i) {
+	if (i->second) {
+	    i->second->silence();
+	    i->second->discardEvents();
+	}
+    }
+
+    m_noteOffs.clear();
+}
+
+void
+AudioGenerator::setTargetChannelCount(size_t targetChannelCount)
+{
+    if (m_targetChannelCount == targetChannelCount) return;
+
+//    std::cerr << "AudioGenerator::setTargetChannelCount(" << targetChannelCount << ")" << std::endl;
+
+    QMutexLocker locker(&m_mutex);
+    m_targetChannelCount = targetChannelCount;
+
+    for (PluginMap::iterator i = m_synthMap.begin(); i != m_synthMap.end(); ++i) {
+	if (i->second) i->second->setIdealChannelCount(targetChannelCount);
+    }
+}
+
+size_t
+AudioGenerator::getBlockSize() const
+{
+    return m_pluginBlockSize;
+}
+
+size_t
+AudioGenerator::mixModel(Model *model, size_t startFrame, size_t frameCount,
+			 float **buffer, size_t fadeIn, size_t fadeOut)
+{
+    if (m_sourceSampleRate == 0) {
+	std::cerr << "WARNING: AudioGenerator::mixModel: No base source sample rate available" << std::endl;
+	return frameCount;
+    }
+
+    QMutexLocker locker(&m_mutex);
+
+    PlayParameters *parameters =
+	PlayParameterRepository::getInstance()->getPlayParameters(model);
+    if (!parameters) return frameCount;
+
+    bool playing = !parameters->isPlayMuted();
+    if (!playing) return frameCount;
+
+    float gain = parameters->getPlayGain();
+    float pan = parameters->getPlayPan();
+
+    DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model);
+    if (dtvm) {
+	return mixDenseTimeValueModel(dtvm, startFrame, frameCount,
+				      buffer, gain, pan, fadeIn, fadeOut);
+    }
+
+    SparseOneDimensionalModel *sodm = dynamic_cast<SparseOneDimensionalModel *>
+	(model);
+    if (sodm) {
+	return mixSparseOneDimensionalModel(sodm, startFrame, frameCount,
+					    buffer, gain, pan, fadeIn, fadeOut);
+    }
+
+    NoteModel *nm = dynamic_cast<NoteModel *>(model);
+    if (nm) {
+	return mixNoteModel(nm, startFrame, frameCount,
+			    buffer, gain, pan, fadeIn, fadeOut);
+    }
+
+    return frameCount;
+}
+
+size_t
+AudioGenerator::mixDenseTimeValueModel(DenseTimeValueModel *dtvm,
+				       size_t startFrame, size_t frames,
+				       float **buffer, float gain, float pan,
+				       size_t fadeIn, size_t fadeOut)
+{
+    static float *channelBuffer = 0;
+    static size_t channelBufSiz = 0;
+
+    size_t totalFrames = frames + fadeIn/2 + fadeOut/2;
+
+    if (channelBufSiz < totalFrames) {
+	delete[] channelBuffer;
+	channelBuffer = new float[totalFrames];
+	channelBufSiz = totalFrames;
+    }
+    
+    size_t got = 0;
+    size_t prevChannel = 999;
+
+    for (size_t c = 0; c < m_targetChannelCount; ++c) {
+
+	size_t sourceChannel = (c % dtvm->getChannelCount());
+
+//	std::cerr << "mixing channel " << c << " from source channel " << sourceChannel << std::endl;
+
+	float channelGain = gain;
+	if (pan != 0.0) {
+	    if (c == 0) {
+		if (pan > 0.0) channelGain *= 1.0 - pan;
+	    } else {
+		if (pan < 0.0) channelGain *= pan + 1.0;
+	    }
+	}
+
+	if (prevChannel != sourceChannel) {
+	    if (startFrame >= fadeIn/2) {
+		got = dtvm->getValues
+		    (sourceChannel,
+		     startFrame - fadeIn/2, startFrame + frames + fadeOut/2,
+		     channelBuffer);
+	    } else {
+		size_t missing = fadeIn/2 - startFrame;
+		got = dtvm->getValues
+		    (sourceChannel,
+		     0, startFrame + frames + fadeOut/2,
+		     channelBuffer + missing);
+	    }	    
+	}
+	prevChannel = sourceChannel;
+
+	for (size_t i = 0; i < fadeIn/2; ++i) {
+	    float *back = buffer[c];
+	    back -= fadeIn/2;
+	    back[i] += (channelGain * channelBuffer[i] * i) / fadeIn;
+	}
+
+	for (size_t i = 0; i < frames + fadeOut/2; ++i) {
+	    float mult = channelGain;
+	    if (i < fadeIn/2) {
+		mult = (mult * i) / fadeIn;
+	    }
+	    if (i > frames - fadeOut/2) {
+		mult = (mult * ((frames + fadeOut/2) - i)) / fadeOut;
+	    }
+	    buffer[c][i] += mult * channelBuffer[i];
+	}
+    }
+
+    return got;
+}
+  
+size_t
+AudioGenerator::mixSparseOneDimensionalModel(SparseOneDimensionalModel *sodm,
+					     size_t startFrame, size_t frames,
+					     float **buffer, float gain, float pan,
+					     size_t /* fadeIn */,
+					     size_t /* fadeOut */)
+{
+    RealTimePluginInstance *plugin = m_synthMap[sodm];
+    if (!plugin) return 0;
+
+    size_t latency = plugin->getLatency();
+    size_t blocks = frames / m_pluginBlockSize;
+    
+    //!!! hang on -- the fact that the audio callback play source's
+    //buffer is a multiple of the plugin's buffer size doesn't mean
+    //that we always get called for a multiple of it here (because it
+    //also depends on the JACK block size).  how should we ensure that
+    //all models write the same amount in to the mix, and that we
+    //always have a multiple of the plugin buffer size?  I guess this
+    //class has to be queryable for the plugin buffer size & the
+    //callback play source has to use that as a multiple for all the
+    //calls to mixModel
+
+    size_t got = blocks * m_pluginBlockSize;
+
+#ifdef DEBUG_AUDIO_GENERATOR
+    std::cout << "mixModel [sparse]: frames " << frames
+	      << ", blocks " << blocks << std::endl;
+#endif
+
+    snd_seq_event_t onEv;
+    onEv.type = SND_SEQ_EVENT_NOTEON;
+    onEv.data.note.channel = 0;
+    onEv.data.note.note = 64;
+    onEv.data.note.velocity = 127;
+
+    snd_seq_event_t offEv;
+    offEv.type = SND_SEQ_EVENT_NOTEOFF;
+    offEv.data.note.channel = 0;
+    offEv.data.note.velocity = 0;
+    
+    NoteOffSet &noteOffs = m_noteOffs[sodm];
+
+    for (size_t i = 0; i < blocks; ++i) {
+
+	size_t reqStart = startFrame + i * m_pluginBlockSize;
+
+	SparseOneDimensionalModel::PointList points =
+	    sodm->getPoints(reqStart + latency,
+			    reqStart + latency + m_pluginBlockSize);
+
+        Vamp::RealTime blockTime = Vamp::RealTime::frame2RealTime
+	    (startFrame + i * m_pluginBlockSize, m_sourceSampleRate);
+
+	for (SparseOneDimensionalModel::PointList::iterator pli =
+		 points.begin(); pli != points.end(); ++pli) {
+
+	    size_t pliFrame = pli->frame;
+
+	    if (pliFrame >= latency) pliFrame -= latency;
+
+	    if (pliFrame < reqStart ||
+		pliFrame >= reqStart + m_pluginBlockSize) continue;
+
+	    while (noteOffs.begin() != noteOffs.end() &&
+		   noteOffs.begin()->frame <= pliFrame) {
+
+                Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime
+		    (noteOffs.begin()->frame, m_sourceSampleRate);
+
+		offEv.data.note.note = noteOffs.begin()->pitch;
+
+#ifdef DEBUG_AUDIO_GENERATOR
+		std::cerr << "mixModel [sparse]: sending note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl;
+#endif
+
+		plugin->sendEvent(eventTime, &offEv);
+		noteOffs.erase(noteOffs.begin());
+	    }
+
+            Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime
+		(pliFrame, m_sourceSampleRate);
+	    
+	    plugin->sendEvent(eventTime, &onEv);
+
+#ifdef DEBUG_AUDIO_GENERATOR
+	    std::cout << "mixModel [sparse]: point at frame " << pliFrame << ", block start " << (startFrame + i * m_pluginBlockSize) << ", resulting time " << eventTime << std::endl;
+#endif
+	    
+	    size_t duration = 7000; // frames [for now]
+	    NoteOff noff;
+	    noff.pitch = onEv.data.note.note;
+	    noff.frame = pliFrame + duration;
+	    noteOffs.insert(noff);
+	}
+
+	while (noteOffs.begin() != noteOffs.end() &&
+	       noteOffs.begin()->frame <=
+	       startFrame + i * m_pluginBlockSize + m_pluginBlockSize) {
+
+            Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime
+		(noteOffs.begin()->frame, m_sourceSampleRate);
+
+	    offEv.data.note.note = noteOffs.begin()->pitch;
+
+#ifdef DEBUG_AUDIO_GENERATOR
+		std::cerr << "mixModel [sparse]: sending leftover note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl;
+#endif
+
+	    plugin->sendEvent(eventTime, &offEv);
+	    noteOffs.erase(noteOffs.begin());
+	}
+	
+	plugin->run(blockTime);
+	float **outs = plugin->getAudioOutputBuffers();
+
+	for (size_t c = 0; c < m_targetChannelCount; ++c) {
+#ifdef DEBUG_AUDIO_GENERATOR
+	    std::cout << "mixModel [sparse]: adding " << m_pluginBlockSize << " samples from plugin output " << c << std::endl;
+#endif
+
+	    size_t sourceChannel = (c % plugin->getAudioOutputCount());
+
+	    float channelGain = gain;
+	    if (pan != 0.0) {
+		if (c == 0) {
+		    if (pan > 0.0) channelGain *= 1.0 - pan;
+		} else {
+		    if (pan < 0.0) channelGain *= pan + 1.0;
+		}
+	    }
+
+	    for (size_t j = 0; j < m_pluginBlockSize; ++j) {
+		buffer[c][i * m_pluginBlockSize + j] +=
+		    channelGain * outs[sourceChannel][j];
+	    }
+	}
+    }
+
+    return got;
+}
+
+    
+//!!! mucho duplication with above -- refactor
+size_t
+AudioGenerator::mixNoteModel(NoteModel *nm,
+			     size_t startFrame, size_t frames,
+			     float **buffer, float gain, float pan,
+			     size_t /* fadeIn */,
+			     size_t /* fadeOut */)
+{
+    RealTimePluginInstance *plugin = m_synthMap[nm];
+    if (!plugin) return 0;
+
+    size_t latency = plugin->getLatency();
+    size_t blocks = frames / m_pluginBlockSize;
+    
+    //!!! hang on -- the fact that the audio callback play source's
+    //buffer is a multiple of the plugin's buffer size doesn't mean
+    //that we always get called for a multiple of it here (because it
+    //also depends on the JACK block size).  how should we ensure that
+    //all models write the same amount in to the mix, and that we
+    //always have a multiple of the plugin buffer size?  I guess this
+    //class has to be queryable for the plugin buffer size & the
+    //callback play source has to use that as a multiple for all the
+    //calls to mixModel
+
+    size_t got = blocks * m_pluginBlockSize;
+
+#ifdef DEBUG_AUDIO_GENERATOR
+    std::cout << "mixModel [note]: frames " << frames
+	      << ", blocks " << blocks << std::endl;
+#endif
+
+    snd_seq_event_t onEv;
+    onEv.type = SND_SEQ_EVENT_NOTEON;
+    onEv.data.note.channel = 0;
+    onEv.data.note.note = 64;
+    onEv.data.note.velocity = 127;
+
+    snd_seq_event_t offEv;
+    offEv.type = SND_SEQ_EVENT_NOTEOFF;
+    offEv.data.note.channel = 0;
+    offEv.data.note.velocity = 0;
+    
+    NoteOffSet &noteOffs = m_noteOffs[nm];
+
+    for (size_t i = 0; i < blocks; ++i) {
+
+	size_t reqStart = startFrame + i * m_pluginBlockSize;
+
+	NoteModel::PointList points =
+	    nm->getPoints(reqStart + latency,
+			    reqStart + latency + m_pluginBlockSize);
+
+        Vamp::RealTime blockTime = Vamp::RealTime::frame2RealTime
+	    (startFrame + i * m_pluginBlockSize, m_sourceSampleRate);
+
+	for (NoteModel::PointList::iterator pli =
+		 points.begin(); pli != points.end(); ++pli) {
+
+	    size_t pliFrame = pli->frame;
+
+	    if (pliFrame >= latency) pliFrame -= latency;
+
+	    if (pliFrame < reqStart ||
+		pliFrame >= reqStart + m_pluginBlockSize) continue;
+
+	    while (noteOffs.begin() != noteOffs.end() &&
+		   noteOffs.begin()->frame <= pliFrame) {
+
+                Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime
+		    (noteOffs.begin()->frame, m_sourceSampleRate);
+
+		offEv.data.note.note = noteOffs.begin()->pitch;
+
+#ifdef DEBUG_AUDIO_GENERATOR
+		std::cerr << "mixModel [note]: sending note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl;
+#endif
+
+		plugin->sendEvent(eventTime, &offEv);
+		noteOffs.erase(noteOffs.begin());
+	    }
+
+            Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime
+		(pliFrame, m_sourceSampleRate);
+	    
+            if (nm->getScaleUnits() == "Hz") {
+                onEv.data.note.note = Pitch::getPitchForFrequency(pli->value);
+            } else {
+                onEv.data.note.note = lrintf(pli->value);
+            }
+
+	    plugin->sendEvent(eventTime, &onEv);
+
+#ifdef DEBUG_AUDIO_GENERATOR
+	    std::cout << "mixModel [note]: point at frame " << pliFrame << ", block start " << (startFrame + i * m_pluginBlockSize) << ", resulting time " << eventTime << std::endl;
+#endif
+	    
+	    size_t duration = pli->duration;
+            if (duration == 0 || duration == 1) {
+                duration = m_sourceSampleRate / 20;
+            }
+	    NoteOff noff;
+	    noff.pitch = onEv.data.note.note;
+	    noff.frame = pliFrame + duration;
+	    noteOffs.insert(noff);
+	}
+
+	while (noteOffs.begin() != noteOffs.end() &&
+	       noteOffs.begin()->frame <=
+	       startFrame + i * m_pluginBlockSize + m_pluginBlockSize) {
+
+            Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime
+		(noteOffs.begin()->frame, m_sourceSampleRate);
+
+	    offEv.data.note.note = noteOffs.begin()->pitch;
+
+#ifdef DEBUG_AUDIO_GENERATOR
+		std::cerr << "mixModel [note]: sending leftover note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl;
+#endif
+
+	    plugin->sendEvent(eventTime, &offEv);
+	    noteOffs.erase(noteOffs.begin());
+	}
+	
+	plugin->run(blockTime);
+	float **outs = plugin->getAudioOutputBuffers();
+
+	for (size_t c = 0; c < m_targetChannelCount; ++c) {
+#ifdef DEBUG_AUDIO_GENERATOR
+	    std::cout << "mixModel [note]: adding " << m_pluginBlockSize << " samples from plugin output " << c << std::endl;
+#endif
+
+	    size_t sourceChannel = (c % plugin->getAudioOutputCount());
+
+	    float channelGain = gain;
+	    if (pan != 0.0) {
+		if (c == 0) {
+		    if (pan > 0.0) channelGain *= 1.0 - pan;
+		} else {
+		    if (pan < 0.0) channelGain *= pan + 1.0;
+		}
+	    }
+
+	    for (size_t j = 0; j < m_pluginBlockSize; ++j) {
+		buffer[c][i * m_pluginBlockSize + j] += 
+		    channelGain * outs[sourceChannel][j];
+	    }
+	}
+    }
+
+    return got;
+}
+